Add mute state field to AudioFrame.

webrtc/voice_engine/utility_unittest.cc

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 29 matching lines @@
   PushResampler<int16_t> resampler_;
   AudioFrame src_frame_;
   AudioFrame dst_frame_;
   AudioFrame golden_frame_;
 };
 
 // Sets the signal value to increase by |data| with every sample. Floats are
 // used so non-integer values result in rounding error, but not an accumulating
 // error.
 void SetMonoFrame(float data, int sample_rate_hz, AudioFrame* frame) {
-  memset(frame->data_, 0, sizeof(frame->data_));
+  frame->Mute();
   frame->num_channels_ = 1;
   frame->sample_rate_hz_ = sample_rate_hz;
   frame->samples_per_channel_ = rtc::CheckedDivExact(sample_rate_hz, 100);
+  int16_t* frame_data = frame->mutable_data();
   for (size_t i = 0; i < frame->samples_per_channel_; i++) {
-    frame->data_[i] = static_cast<int16_t>(data * i);
+    frame_data[i] = static_cast<int16_t>(data * i);
   }
 }
 
 // Keep the existing sample rate.
 void SetMonoFrame(float data, AudioFrame* frame) {
   SetMonoFrame(data, frame->sample_rate_hz_, frame);
 }
 
 // Sets the signal value to increase by |left| and |right| with every sample in
 // each channel respectively.
 void SetStereoFrame(float left,
                     float right,
                     int sample_rate_hz,
                     AudioFrame* frame) {
-  memset(frame->data_, 0, sizeof(frame->data_));
+  frame->Mute();
   frame->num_channels_ = 2;
   frame->sample_rate_hz_ = sample_rate_hz;
   frame->samples_per_channel_ = rtc::CheckedDivExact(sample_rate_hz, 100);
+  int16_t* frame_data = frame->mutable_data();
   for (size_t i = 0; i < frame->samples_per_channel_; i++) {
-    frame->data_[i * 2] = static_cast<int16_t>(left * i);
-    frame->data_[i * 2 + 1] = static_cast<int16_t>(right * i);
+    frame_data[i * 2] = static_cast<int16_t>(left * i);
+    frame_data[i * 2 + 1] = static_cast<int16_t>(right * i);
   }
 }
 
 // Keep the existing sample rate.
 void SetStereoFrame(float left, float right, AudioFrame* frame) {
   SetStereoFrame(left, right, frame->sample_rate_hz_, frame);
 }
 
 // Sets the signal value to increase by |ch1|, |ch2|, |ch3|, |ch4| with every
 // sample in each channel respectively.
 void SetQuadFrame(float ch1,
                   float ch2,
                   float ch3,
                   float ch4,
                   int sample_rate_hz,
                   AudioFrame* frame) {
-  memset(frame->data_, 0, sizeof(frame->data_));
+  frame->Mute();
   frame->num_channels_ = 4;
   frame->sample_rate_hz_ = sample_rate_hz;
   frame->samples_per_channel_ = rtc::CheckedDivExact(sample_rate_hz, 100);
+  int16_t* frame_data = frame->mutable_data();
   for (size_t i = 0; i < frame->samples_per_channel_; i++) {
-    frame->data_[i * 4] = static_cast<int16_t>(ch1 * i);
-    frame->data_[i * 4 + 1] = static_cast<int16_t>(ch2 * i);
-    frame->data_[i * 4 + 2] = static_cast<int16_t>(ch3 * i);
-    frame->data_[i * 4 + 3] = static_cast<int16_t>(ch4 * i);
+    frame_data[i * 4] = static_cast<int16_t>(ch1 * i);
+    frame_data[i * 4 + 1] = static_cast<int16_t>(ch2 * i);
+    frame_data[i * 4 + 2] = static_cast<int16_t>(ch3 * i);
+    frame_data[i * 4 + 3] = static_cast<int16_t>(ch4 * i);
   }
 }
 
 void VerifyParams(const AudioFrame& ref_frame, const AudioFrame& test_frame) {
   EXPECT_EQ(ref_frame.num_channels_, test_frame.num_channels_);
   EXPECT_EQ(ref_frame.samples_per_channel_, test_frame.samples_per_channel_);
   EXPECT_EQ(ref_frame.sample_rate_hz_, test_frame.sample_rate_hz_);
 }
 
 // Computes the best SNR based on the error between |ref_frame| and
 // |test_frame|. It allows for up to a |max_delay| in samples between the
 // signals to compensate for the resampling delay.
 float ComputeSNR(const AudioFrame& ref_frame, const AudioFrame& test_frame,
                  size_t max_delay) {
   VerifyParams(ref_frame, test_frame);
   float best_snr = 0;
   size_t best_delay = 0;
   for (size_t delay = 0; delay <= max_delay; delay++) {
     float mse = 0;
     float variance = 0;
+    const int16_t* ref_frame_data = ref_frame.data();
+    const int16_t* test_frame_data = test_frame.data();
     for (size_t i = 0; i < ref_frame.samples_per_channel_ *
         ref_frame.num_channels_ - delay; i++) {
-      int error = ref_frame.data_[i] - test_frame.data_[i + delay];
+      int error = ref_frame_data[i] - test_frame_data[i + delay];
       mse += error * error;
-      variance += ref_frame.data_[i] * ref_frame.data_[i];
+      variance += ref_frame_data[i] * ref_frame_data[i];
     }
     float snr = 100;  // We assign 100 dB to the zero-error case.
     if (mse > 0)
       snr = 10 * log10(variance / mse);
     if (snr > best_snr) {
       best_snr = snr;
       best_delay = delay;
     }
   }
   printf("SNR=%.1f dB at delay=%" PRIuS "\n", best_snr, best_delay);
   return best_snr;
 }
 
 void VerifyFramesAreEqual(const AudioFrame& ref_frame,
                           const AudioFrame& test_frame) {
   VerifyParams(ref_frame, test_frame);
+  const int16_t* ref_frame_data = ref_frame.data();
+  const int16_t* test_frame_data  = test_frame.data();
   for (size_t i = 0;
        i < ref_frame.samples_per_channel_ * ref_frame.num_channels_; i++) {
-    EXPECT_EQ(ref_frame.data_[i], test_frame.data_[i]);
+    EXPECT_EQ(ref_frame_data[i], test_frame_data[i]);
   }
 }
 
 void UtilityTest::RunResampleTest(int src_channels,
                                   int src_sample_rate_hz,
                                   int dst_channels,
                                   int dst_sample_rate_hz) {
   PushResampler<int16_t> resampler;  // Create a new one with every test.
   const int16_t kSrcCh1 = 30;  // Shouldn't overflow for any used sample rate.
   const int16_t kSrcCh2 = 15;
@@ skipping 104 matching lines @@
                           kDstChannels[dst_channel], kSampleRates[dst_rate]);
         }
       }
     }
   }
 }
 
 }  // namespace
 }  // namespace voe
 }  // namespace webrtc